On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Try this:
modprobe zaptel
modprobe your card driver, like wctdm
We ran into the same thing, and the only way I can get it to work
(which is goofy, but it does work) is modprobing the same device
multiple times.
So
modprobe zaptel
modprobe
Hello,
I have found and read Steven Critchfields writeup on how to use ZapRas
(Thanks Critch!), however I am a bit confused.
His write up is here:
http://copilotconsulting.com/mail-archives/asterisk.2003/msg01030.html
Currently we have a full PRI (23B Channels, 1D) coming into our
Asterisk Box
Just a point of reference for others, we had a lot of echo on an Adtran Channel Bank connected to a T100P.
The end poitns were all Polycom Ip500's on a local lan connecting to *.
We couldn't solve the echo with any software settings and installed a
Tellabs Echo Can shelf 255D i believe with 64ms
I have the same problem, after about a month the card doesn't report anyincoming calls anymore to the console. Don't know the rev of my card yet,
unloading asterisk and unloading the modules and then restartingeverything does seem to help though, no need to
On 10/4/05, 1 2 [EMAIL PROTECTED] wrote:
HiHypothetical but quite possible scenerio:Attempted emergency 911 call but all zap channels are already in use.
Is there any way to hangup zap channels before dial(Zap/g1/911) or equivalent.AFAIK hangup doesn't except options so I CANNOT do something like
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of VenkataRao Chimata
Sent: Saturday, January 15, 2005 12:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to create channel of type 'Zap'
Hi friends
I
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of VenkataRao Chimata
Sent: Saturday, January 15, 2005 12:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to create channel of type 'Zap'
Hi friends
I
1. When it dials out, many times the digits are not properly recognized
by telco as I hear the announcement please check the number and dial
again although I see on the screen that the dialed number is correct.
Had the same problem with an older central office and the 'w' fixed it.
I
Hello,
I was curious how people had
the timing setup for their T100P and Total Access 750.
We have been getting Red
Alarms once a day for 5 seconds. I think the line is losing sync and
resyncing.
Currently, it is set like
this:
span=1,1,0,esf,b8zs
Should I set the channel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TC
Sent: Thursday, December 23, 2004 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RedAlarm (t100p - Adtran Total Access 750)
Why would you say channel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Wednesday, December 22, 2004 2:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] MWI not working on Polycom Phones
Hi All -
I'm running version
Hello,
About every 2 or 3 days I notice in the messages log file:
Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 2: Red Alarm
Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 3: Red Alarm
Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 4: Red Alarm
Dec 17 08:39:27
I think it is 456
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Darnell
Sent: Friday, December 10, 2004 1:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Lost Password to
Brent,
I am embarased to say that I changed it from 456. Can't seem to find
the paper it was written on! :(
-Matt
Hi Matt,
Sorry I read your last message too quickly. There is an admin guide at
http://www.freedomphones.net/polycom/files/Admin_Guide-SoundPoint_IP_SIP
_2004-06-16.pdf
Hello,
We have a high volume of incoming and outgoing calls that come in via
our analog POTS lines connected to FXO cards in an Adtran TA750. This
is connected to a T100P.
We are using Polycom IP 500's. The problem we are experiencing is, on
frequent occasions, when someone dials out, there is
I have noticed some echo when using the handset/headset, but I will
take
another look at the config files today. I noticed that 'ec' which I
assume is echo cancellation is disabled by default on both the
handset
and the headset. I also noticed the nearby text which said Don't
touch
these
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Caley
Sent: Wednesday, November 17, 2004 9:14 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Port for Asterisk
I set an Asterisk server, what ports would I need to open for
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Richard Reina
Sent: Wednesday, November 10, 2004 3:27 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Hooking up a an Adit 600
I have purchased an Adit 600 but with 6 FXS 8 channel
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Leah Newmark
Sent: Monday, November 01, 2004 10:08 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicemail with separate greetings based
onextension
Is there a way to set up a
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, November 01, 2004 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: How far is IAX to be a Standard
No
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Scott Laird
Sent: Saturday, October 23, 2004 12:37 PM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] Cheap hosted servers and Asterisk
Does anyone have any experience with
Hello,
Our client currently has two X100P's running in an HP box that has been
running for almost a year now with no problems. They have found however
that two phone lines are not enough and are bringing in a third phone
line. I wouldn't expect this line to be used very often as there are
only
See comments inline...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stewart M. Ives
Sent: Friday, October 15, 2004 12:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] New Project - IP Phone Sources
Question: If I just want
Hi Joe,
The Polycom IP phones support this, however currently there is no
support for it in *.
I don't think the SIP RFC requires support for this.
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joe Greco
Sent: Wednesday,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mark C. Thomas
Subject: Re: [Asterisk-Users] Red Alarm on X100P
I haven't tried disconnecting the phone line, I'll try that
next time.
If it was a co problem, I wouldn't think
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of el Flynn
Sent: Wednesday, September 22, 2004 7:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Transfering incoming calls using same
line
Hello,
I am receiving an error in my error logs any time I receive a call on
the third line in our hunt group.
Sep 20 13:15:03 WARNING[1116939584]: Ring/Off-hook in strange state 6 on
channel 3
The weird part is that the calls seem to work fine, just this error
message is logged. Currently, I
Hello,
Has anyone else ever experienced Static Problems with a T100P connected
to an Adtran Total Access 750? We have two FXO modules in the Chasis to
interface to Verizon. At first I thought it was just one line, and was
Verizon's fault, but now we are seeing it across all lines and it comes
Rodolfo,
Did you do a modprobe wcfxo
Also, did you perform a ztcfg - after the modprobe? What do the
two report? Additionally, what does cat /proc/interrupts report?
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Marconi Rivello
Sent: Friday, September 10, 2004 1:47 PM
To: Asterisk
Subject: [Asterisk-Users] Red Alarm
I made some progress... I was looking for an indication in the system
that
Hello,
I recently upgraded to Sip 1.3.1 and noticed that the Reject Button is
no longer appearent on the screen when a second incoming call comes in
unless I press the hold button on the first call.
Does anyone have a work around for this to reject a call while
continuing to talk to the first
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joe Antkowiak
Sent: Wednesday, September 08, 2004 3:36 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] successful echo cancellation!!! (multitech)
We recently had a customer
That website is not owned by the providers of this mailing list, as
far as I know. The WIKI is an independently-owned resource where
anyone can post anything (on topic) they like. There's no screening
except for the fact that anyone can also remove, change or comment
on anything they don't
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Greg Blakely
Sent: Thursday, September 02, 2004 2:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Hard Ground (On Ring)
If you have a
Just out of curiosity,
What version of CVS and Polycom SIP software are you running happily?
Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0?
I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with
poor results. Transferring did not work as expected. Using the # key to
do
Look up the word persist in the XML config file...
- Brent
On Tue, 31 Aug 2004, Reid A. Forrest wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Marlowe
Sent: Monday, August 30, 2004 12:55 PM
To: [EMAIL PROTECTED]
I don't have echo problems on my X100P (at home) but that won't stop
me from dumping it in favour of a Sipura SPA-3000 next month, once it
gets full UK support in firmware (caller ID etc.).
It may not be a big deal, but other considerations are:
There is another box to manager.
Another
Hello,
We keep having a really bad static problem on phone calls completed
using a Adtran TA750 and T100P card. The phones are Polycom IP 500
phones and, it occurs across all phones. Not just one.
Everything appears to be on it's own interrupt. I noticed the last time
we did this, we rewired
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig
Neumanns
Sent: Tuesday, August 24, 2004 5:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk with Adit 600
Hello,
I am connecting Asterisk to an Adit 600 via a T100P. Unfortunately I
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey
Richey
Sent: Wednesday, August 25, 2004 12:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] WTS: Just arrived Brand New Cisco IP phones
Hello
I WTS (WANT TO SELL) THE FOLLOWING CISCO
Hi,
I have a problem with a Digium quad E1 card. It seems when I make
outgoing calls to any party, when that person talks on the line, they
hear scratching and static (there's also background static, but less of
it). The person making the call from asterisk (via the E1) doesn't hear
any of
I am currently a new asterisk user I have worked with the old rolm systems
in the past. I have been asked to look around and find out how to do a few
things in asterisk, either in asterisk itself or with third party software.
The features that I am looking for are:
1. A good management
On Fri, 6 Aug 2004, Marc C Storck wrote:
I'm looking for a similar feature...
Dial a number via ZAP/g1
after the line gets answered
wait 10 seconds
send DTMF
Regards,
Marc
Hey Marc,
Check out the WiKi. It has a lot of useful commands that can help you
out, plus you won't have to
On Thu, 5 Aug 2004, Andrew Newton wrote:
Our company has several sites. Two of which we regularly make calls between over
the PSTN. Both sites currently have thier own PBX systems but they are not
linked by any means other than the PSTN (and we pay by the min for calls
between sites)
Both
Hi,
what are the Systemrequirements for Asterisk with SIP?
Moritz Beierlein
Hi Mortiz,
The system requirements are not really a matter of Asterisk with Sip.
Posting some more information in regads to number of Sip clients, codec
requirements, number transcoding streams, etc would be more
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Friday, July 30, 2004 6:01 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] New to IP-PBX
If you don't do any transcoding, and turn canreinvite=on for your
Not
Can we keep this type of stuff off the list? It's annoying to get 100's
of emails a day with nothing other than a simple conversation. Please
take it offlist with someone you're worknig with.
Joe
I know, I'll be flamed for this.
Agreed. Maybe an Asterisk-Provider list? Every
Hello,
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is all on an internal switched 100mb lan.
Has
Hello,
I have been researching Echo Can's for a while now, and wanted to post
this out to the list to solicit feedback (and if my assumptions are
correct, hopefully help others out)...
If anyone out there knows anything about wiring up an Echo Can, and my
outline below is incorrect, please let
Hello,
I have been researching Echo Can's for a while now, and wanted to post
this out to the list to solicit feedback (and if my assumptions are
correct, hopefully help others out)...
If anyone out there knows anything about wiring up an Echo Can, and my
outline below is incorrect,
HI ALL;
Is astersik enable to translate between different codecs.
I have couple of SIP-UA , one with (a-law) and the other with (g729), registered
with my astersik box.Can astersik translate between alaw-g729 and vice varsa.
Yes.
Also, Google works pretty good too. A simple Google
This is how I have the SIP extension setup:
[2002]
type=friend
username=2002
secret=mypassword
host=dynamic
context=from-sip
mailbox=2002
nat=yes
qualify=yes
dtmfmode=info
reinvite=no
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
callwaiting=1
Not sure how to
Dear Sirs,
I'm running an Asterisk 0.9.1 in a Fedora Core 2 box.
I installed a X100P card on my box and when i try to load modules i am
rejected.
When i type modprobe zaptel my Fedora respond : FATAL: Module zaptel not
found. . The same uccurs when i type modprobe wcfxo
Perform an
Thank you!
Can you tell me more about the dial plan feature? How do you setup the
correct digitmap?
Check the Administrator's Document. You can find it on the Wiki, under IP
Phones.. Polycom. Did you try to look up the digitmap feature before
sending this post? If not, you should be
Also, in this day of motherboard-integrated NICs (even two or three),
what will happen if the mobo dies and has to be replaced?
The same thing that would happen if the NIC died. IMHO it's a good thing
to tie to the NIC, because the chances of the MOBO dieing is not that
extreme. If it does
I know this has been covered before, but could someone please explain the
benefits to starting asterisk various ways. I am partly posting this too,
to see if my assumptions are correct.
Is call quality affected by starting it differently?
My belief is no. Regardless of how you start it,
Chris,
/etc/sip.conf
for each device add:
canreinvite=no
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris HARIGA
Sent: Monday, July 12, 2004 5:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No voice bet/ ext with Polycom
Hi,
We
Hi Rich,
Thanks for your heads up. See comments below.
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Best think through what you're trying to do here. You have multiple
choices on how to interface * to the traditional pstn world, including
Hello,
After reading the lists and taking reccomendations from TC, I have finally
given up on the echo can built into asterisk. I am sick of hearing
complaints from users, so the money spent on a hardware echo can will be
worth its weight in gold.
I am curious however, about some setup and
Ditech Communications ( http://www.ditechcom.com/ ) has a 2 slot and a
4
slot chassis and you can populate them with just 1 echo cancelling
card if
you like.
They have RJ45 jacks and you have to use a T1 Cross cable on both
sides.
We
Hi Robb,
Thanks for your reply. Have you found that
Hello,
I am cruious what exactly status shows. If I do a sip show peers, I get
this table:
2133/213310.10.60.9 D 255.255.255.255 5060 OK
(95 ms)
2120/212010.10.60.2 D 255.255.255.255 5060 OK
(112 ms)
Now, if I exit asterisk, and ping
On Thu, 1 Jul 2004, Mike Benoit wrote:
Obviously the less I spend the better. But if we have to, a few thousand
more I guess. The problem I have is that this setup is more of a trial
run. Once it works, I'm going to be cloning slightly smaller setups to
9 other cities. But they are pretty
Chris Travers Wrote
Before I tell a customer that this would require custom development I
figured I would ask here.
Does Asterisk support pager notification of new voicemails out of the
box? Or do I need an AGI script to do that?
For our notifications, we just send e-mails as text
Hi everyone,
This one has me baffled
We installed 16 Polycom IP 500 Phones in January on an Asterisk Server:
3.0Ghz
768MB of Ram
30GB Harddrives
10/100/1000 NIC
Our Lan setup is like this:
3 Netgear FS526T Switches
2 10/100/1000 Mbps
Hello,
When I enable SIP debugging I receive:
Peer RTP is at port 10.10.60.16:0
Shouldn't the RTP port be a number between 1 - 2?
- Brent
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
..Jun 29 11:56:33 WARNING[-1084538752]: chan_zap.c:704 zt_open: Unable
to specify channel 1: No such device or address
Jun 29 11:56:33 ERROR[-1084538752]: chan_zap.c:5499 mkintf: Unable to
open channel 1: No such device or address
here = 0, tmp-channel = 1, channel = 1
Jun 29 11:56:33
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Thursday, June 24, 2004 5:01 PM
Be careful with that thought... here's the three lines that were
manually changed for testing purposes only (these would have been
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tor Roberts
Sent: Monday, June 14, 2004 8:22 PM
John,
No, I have not tried 1.2, I did not know it was even out. Can this be
downloaded from Polycom's site? If so, I will try it out.
BF You can try doing different things with it, but I know that I am
currently
BF set to level 3 rather than 5 as default with RedHat.
I checked hdparm googling around, what parameter have you set to 3
instead of 5 ?
Alessio Focardi wrote:
I'm pretty sure this is a confusion.
I think
Of course, right now things
like * do not have an adequate reputation to pick up much of that
business. There is, however, a preparedness there for radical change.
When you are able to purchase support contracts on Asterisk (E.g. Yearly
(not hourly)) * will gain a lot of momentum. There
On Wed, 9 Jun 2004, Alessio Focardi wrote:
Asterisk with one HFC isdn card, using the zaptel driver bristuff
All works ok, but voice coming in/out of the isdn card is out of sync,
squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say
launching xwindows.
Alessio,
When I was having
You deserve a large flaming over not reading for that.
Could it be you have not installed CVS..
Lazy, non reading person, scatch that, you just rated below person.
--
Steven Critchfield [EMAIL PROTECTED]
I think we should create a PHP page that gives a quiz regarding
sure it does, how is this any different from--
I'm having trouble with hardware component X and I can't seem to get
help from the vendor. Does anyone have any suggestions?
-reed
Because most likely this will turn into yet another pissing match from
Nufone to disgruntled customers,
I'm just wondering if I could get all this in one line.
Would dialing via IAX2 help rather then through the zaptel lines?
I have also seen key systems before that will ring your cell phone and
prompt you to press 1 if you would like to accept the call, or press 2 if
you would like to enter
We utilize an X100P on a DSL line provisioned
by Verizon with no problems. Just
make sure you place the filters in the right place and you wont have any
problems.
- Brent
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Blackman
Sent:
I do feel the echo cancellation does need some work.
Currently, other than listening to users, there is no way to benchmark or
trouble shoot echo problems.
We find that 2 to 3 out of every 20 calls will experience echo. While
echo is a problem that naturally occurs from SIP - PSTN and vice
We have three Cisco 7940 SIP phones and 1 POTS phone connected to our
*
server with TDM10B fxs card. Our * server is connected to the pstn
with
3
X100P cards. We have similar echo problems but only on our SIP
phones.
We do not have any echo problems with the POTS phone.
We just
I have the same problem, it appears to be a problem with the echo
canceller. I have elected to install a DSP based T1 echo canceller, on
advice from TC.
Will report on how I make out.
Brent
On Thu, 1 Apr 2004, Justin Carlson wrote:
how do you adjust ?
-Original Message-
From:
Google: Asterisk Calling Card
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, March 29, 2004 11:23 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] pre-paid (new to asterisk, pls don't shoot
Check the wiki
http://voip-info.org
Is there any way the mailing list admin can put this in BIG BOLD LETTERS
near the top when you join the list? WW
- Original Message Follows -
Hello there,I am new to Asterisk. This is my first day on
it. Can someone tell me minimum hardware
TC,
Thanks for your recommendation. Looking at sourcing one now. This is
great news.
As I understand it, you need the card, Chasis, and Power Module, and we
should be up and running?
Thanks,
Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
[EMAIL PROTECTED] wrote:
| Hi,
| The echo problem is the X100P. The hybrid is 'unbalanced',
| and basically what happens is that the outgoing sound signal
| comes right-on back as an incoming signal. The reason you
| don't notice it using the TDM400P is that the incoming sound
| is
Wes,
Please let us know how you make out with this.
I experience the same issues.
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wes Marderness
Sent: Monday, March 01, 2004 9:24 AM
To: [EMAIL PROTECTED]
Subject: RE:
We implemented 21 Polycom IP 500 SIP phones in December and the voice
quality with Ulaw is very good. We also use g729 and IAX to connect two
sites together on DSL and voice quality sounds pretty good as well. No
one has complained yet.
The speaker phone is pretty good, and occasionally on POTS
Message -
From: Brent Franks [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 10:21 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
Wim, I made some changes to your Zapata.conf and zaptel.conf config
files below.
Hope
: Re: [Asterisk-Users] Unable to create channem of type
'Zap'
Thanks for the help !
Made changes, still the same message.
I have two NIC's with IRQ 11
The FXO card has IRQ10 (and no other card has IRQ10)
Wim
- Original Message -
From: Brent Franks [EMAIL
Hello,
I was curious if there was any way to play a tone on Attended transfer
once it bridges the party being transferred to the destination?
Basically what is happening now is:
1.) A caller calls in using a zap channel
2.) Call is sent to SIP Polycom Phone - Receptionist
3.) Receptionist
Make sure you run a ztcfg after you do a modprobe.
ztcfg will configure (or bring up) the zap channels on zaptel interface
cards. Do this before starting * and after the modprobe.
(You may also do a ztcfg -v to see whats configured)
- Brent
-Original Message-
From: [EMAIL PROTECTED]
Wim, I made some changes to your Zapata.conf and zaptel.conf config
files below.
Hope this helps.
Also, do a less /proc/interrupts and see if the card is on it's own IRQ.
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
Sent:
Michael,
My polycom phones (SIP) have configuration settings for pattern
matching.
Basically what the phone does is looks for a certain pattern, and once
it is met it transmits the digits to *.
Not sure about the Cisco's as I haven't used them yet, but I'm sure they
support this feature as
Would it be possible here to make the echo canceler option (2,3, etc.) a
configuration file setting, maybe in Zapata rather than compile time?
What does this do to system resources?
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
If you do set busydetect=no and it fixes the problem, try reinstating
busydetect=yes and then put busycount=10.
If the hang-ups persist, try increasing the busy count.
- Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stuart
.G,
Come on now, give us more.
How many concurrent calls? What's your idea of a modern PC? Processor
Speed, HardDrive space, etc? How many voicemail users? Nat'd SIP
clients? What's your lan setup? Codec Translation? Voice Compression,
Echo Cancellation..?
Before you get information, you
They moved to a different location
yesterday.
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson
Sent: Saturday, February 14, 2004
10:07 AM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Digium connectivity issue?
I am having a problem with the logger stopping at random points. It will
then work again randomly, and then stop again.
For example I know warnings have popped up, but they do not go in the
messages file. I see them occur on the console and they are never written
to file. I then notice logs
You will need to set priorities for each one.
For example:
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Playback(pstnallbusy)
exten =
_91NXXNXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
exten = _91NXXNXX,4,Congestion
Basically what happens
Hey John, See my Responses below:
Basically, I'm trying to run 10 POTS lines into the Adtran and get
output into the T100P. Clearly, I have very little idea of what I'm
doing, so I thought I'd ask:
Was in the same position two weeks ago, but have a working system now
tahnks to all the help
On Mon, 2004-02-02 at 11:21, Senad Jordanovic wrote:
Philipp von Klitzing wrote:
Hi!
I would add:
reinvite=no in addition to canreinvite=no.
It may do the trick.
There is no such parameter as reinvite=. Use canreinvite= only.
Well...
Googling... Few months ago produced that
http://www-306.ibm.com/e-business/doc/content/lp/prodigy.html?P_Site=S90
Linux, the Future is Open.
An IBM Commercial shown with the Child Prodigy, it's not the first time
they've shown it.
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Christian,
You can change channel.c source code to be more forgiving of
unrecognized DTMF tones.
Look for my addition near the bottom of this struct:
else if (digit == 'f');
Basically I altered channel.c to this:
static int do_senddigit(struct ast_channel *chan, char digit)
{
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